Need Help

[kashutu]

Hi,
I am new to vicidial, and i am trying to fulfill the requirement in order to post my problem at the forum. I already made one post, this will be mine second post and after 5 days i would be able to post regularly.

Specs:
Goautodial 2.0
Eyebeam
Pentium 4, 2.8GHz with HT
512MB RAM
SIP: Nextiva

I am having problems with an error

Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent.

when i try to manully dial a number and after 10 secs, the message appears.
Any help would be appreciated. I cannot post my asterisk CLI because i am not allwed as yet.

Thanks

[williamconley]

Where do you see this error? (on your soft phone? ... if so, your soft phone likely has not registered to your server successfully).

Please post your Vicidial Version with Build (from the bottom left corner of most Administrator Screens) and where you are in the tutorial. Are you finished with the GoAutoDial "Getting Started" guide? Or have you begun the Vicidial Manager's Manual getting started tutorial?

Asterisk version isn't particularly important as yet, but the way you said that makes it sound like you cannot log in to the vicidial server ...?

[kashutu]

Hi,
I am new to this forum and Goautodial as well. So, if i make any mistakes asking questions, my apologies.

Specs:
Goautodial 2.0
Eyebeam
Pentium 4, 2.8GHz with HT
512MB RAM
SIP: Nextiva

I have recently installed Goautodial 2.0 and configured everything as mentioned in the guide. However, when i log in the agent into eyebeam, i hear "you are the only one in the conference". After than i try to make a manual call, but after 10 secs, LIVE CALL goes green, and immediatley after that i receive the message

DIAL ALERT:

Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent.

Can you please help figure out what am i doing wrong?

Account Entry:
[iBrain01]
disallow=all
allow=ulaw
username=xxx
fromuser=xxx
type=friend
secret=xxx
qualify=no
maxexpirey=3600
host=208.73.146.95
fromdomain=208.73.146.95
insecure=invite
dtmfmode=rfc2833
defaultexpirey=60
nat=yes
canreinvite=no
context=from-trunk


Global String:

SIPtrunk=SIP/iBrain01

Dialplan Entery:

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup
[b]

[williamconley]

Please post your Vicidial Version with Build (from the bottom left corner of most Administrator Screens)

I do not see a registration string entry noted. Do you have one? Does your SIP trunk register?

And you DO need to post your vicidial version with build as noted earlier. That is a requirement for posting (and considering how easy it is to find ... and how FREE the software is ... ).

[boybawang]

first try to disable your firewall by doing service iptables stop on the linux command line.

second ping the ip address of your voice provider from the linux command line ping 208.73.146.95


edit the carrier with this setting:

[ibrain1]
host=208.73.146.95
type=friend
disallow=all
allow=ulaw
username=xxx
fromuser=xxx
secret=xxx
fromdomain=208.73.146.95
insecure=invite
dtmfmode=rfc2833
nat=yes
canreinvite=no
qualify=10000
context=trunkinbound



also try this dialplan for the meantime:


exten => _9.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9.,2,Dial(SIP/ibrain1/${EXTEN},,To)
exten => _9.,3,Hangup


go to the asterisk command line using asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv

note that the 'v' stands for verbosity, alot of v means it will have more verbose output.

verify that the carriers ip is up on the asterisk cli by typing: sip show peers
it would show the carriers latency in ms

make a test call from the softphone thats logged in and check the command line and post the details here

[kashutu]

to williamconley:
Here is the vicidial version:

VERSION: 2.2.1-237
BUILD: 100510-2015

Also the registering string looks like this:

register => xxx:xxx@208.73.146.95:5060/xxx

to boybawang
I did make changes as u described and here is the output:

ibbs*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
gs102/gs102 (Unspecified) D N 0 UNKNOWN
cc150/cc150 (Unspecified) D N 0 UNKNOWN
cc149/cc149 (Unspecified) D N 0 UNKNOWN
cc148/cc148 (Unspecified) D N 0 UNKNOWN
cc147/cc147 (Unspecified) D N 0 UNKNOWN
cc146/cc146 (Unspecified) D N 0 UNKNOWN
cc145/cc145 (Unspecified) D N 0 UNKNOWN
cc144/cc144 (Unspecified) D N 0 UNKNOWN
cc143/cc143 (Unspecified) D N 0 UNKNOWN
cc142/cc142 (Unspecified) D N 0 UNKNOWN
cc141/cc141 (Unspecified) D N 0 UNKNOWN
cc140/cc140 (Unspecified) D N 0 UNKNOWN
cc139/cc139 (Unspecified) D N 0 UNKNOWN
cc138/cc138 (Unspecified) D N 0 UNKNOWN
cc137/cc137 (Unspecified) D N 0 UNKNOWN
cc136/cc136 (Unspecified) D N 0 UNKNOWN
cc135/cc135 (Unspecified) D N 0 UNKNOWN
cc134/cc134 (Unspecified) D N 0 UNKNOWN
cc133/cc133 (Unspecified) D N 0 UNKNOWN
cc132/cc132 (Unspecified) D N 0 UNKNOWN
cc131/cc131 (Unspecified) D N 0 UNKNOWN
cc130/cc130 (Unspecified) D N 0 UNKNOWN
cc129/cc129 (Unspecified) D N 0 UNKNOWN
cc128/cc128 (Unspecified) D N 0 UNKNOWN
cc127/cc127 (Unspecified) D N 0 UNKNOWN
cc126/cc126 (Unspecified) D N 0 UNKNOWN
cc125/cc125 (Unspecified) D N 0 UNKNOWN
cc124/cc124 (Unspecified) D N 0 UNKNOWN
cc123/cc123 (Unspecified) D N 0 UNKNOWN
cc122/cc122 (Unspecified) D N 0 UNKNOWN
cc121/cc121 (Unspecified) D N 0 UNKNOWN
cc120/cc120 (Unspecified) D N 0 UNKNOWN
cc119/cc119 (Unspecified) D N 0 UNKNOWN
cc118/cc118 (Unspecified) D N 0 UNKNOWN
cc117/cc117 (Unspecified) D N 0 UNKNOWN
cc116/cc116 (Unspecified) D N 0 UNKNOWN
cc115/cc115 (Unspecified) D N 0 UNKNOWN
cc114/cc114 (Unspecified) D N 0 UNKNOWN
cc113/cc113 (Unspecified) D N 0 UNKNOWN
cc112/cc112 (Unspecified) D N 0 UNKNOWN
cc111/cc111 (Unspecified) D N 0 UNKNOWN
cc110/cc110 (Unspecified) D N 0 UNKNOWN
cc109/cc109 (Unspecified) D N 0 UNKNOWN
cc108/cc108 (Unspecified) D N 0 UNKNOWN
cc107/cc107 (Unspecified) D N 0 UNKNOWN
cc106/cc106 (Unspecified) D N 0 UNKNOWN
cc105/cc105 (Unspecified) D N 0 UNKNOWN
cc104/cc104 (Unspecified) D N 0 UNKNOWN
cc103/cc103 (Unspecified) D N 0 UNKNOWN
cc102/cc102 (Unspecified) D N 0 UNKNOWN
cc101/cc101 (Unspecified) D N 0 UNKNOWN
cc100/cc100 (Unspecified) D N 0 UNKNOWN
ibrain01/324488996 (Unspecified) D N 0 UNKNOWN
53 sip peers [Monitored: 0 online, 53 offline Unmonitored: 0 online, 0 offline]
ibbs*CLI> sip show registry
Host Username Refresh State Reg.Time
208.73.146.95:5060 324488996 61 Registered Sun, 17 Apr 2011 17:30:04



Not sure if it is getting registered or not.

[boybawang]

did you try applying the changes I suggested?

[kashutu]

Yes i did apply all the changes. I can see it is getting registered and also i can see it under Sip SHow Peers as well.

After looking into CLI, this is the WARNING message i get

WARNING[24940]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Apr 17 19:00:51] == Everyone is busy/congested at this time (1:0/0/1)

Also, i can see the same message when i try to make a manual call from Agent's screen.

DIAL ALERT:

Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent.

[williamconley]

post your new carrier settings (all of them at once, so we don't assume anything from an earlier posting that may have changed).

it would appear that your ibrain is not spelled the same in two places. the system cannot create the sip connection for a reason, this is often the reason (ie: iBrain is not the same as ibrain or iBrain01 vs iBrain1)

[kashutu]

Here is the Carrier Settings:

Registration String:
register => xxx:xxx@208.73.146.95:5060/xxx

Account Entry:
[ibrain01]
host=208.73.146.95
type=friend
disallow=all
allow=ulaw
username=xxx
fromuser=xxx
secret=xxx
fromdomain=208.73.146.95
insecure=invite
dtmfmode=rfc2833
nat=yes
canreinvite=no
qualify=10000
context=trunkinbound

Dialplan Entry:
exten => _9.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9.,2,Dial(SIP/ibrain01/${EXTEN},,To)
exten => _9.,3,Hangup

Let me know if you require anything else

Thanks

[williamconley]

If dialing in the US, use these:

Global String:

SIPtrunk=SIP/ibrain01

Dialplan Entry:

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup

Note that the ":1" in EXTEN:1 removes the "9" at the beginning of your dial string. Without it, the 9 will be sent and all numbers sent to your carrier will begin with 9 (unless all the numbers you are dialing begin with 9, that may not have great results).

Also note that the global string, while not required, causes your dialplan entry to be "uniform", thus easily checked for problems AND eases changing between carriers and even alternate complex dialplans which no longer need the actual carrier in them, just a variable. We often use "Carrier1" or "DIAL9TRUNK" or similar for the global string, thus making their relation to the others fairly obvious (since one cannot run a call center with a SINGLE carrier ... too much risk involved, you should have at least three).

If you are NOT dialing in the US, share the pattern construct information with us and we can build you one (dial prefix from campaign, plus dial code which may or may not include a country code, plus phone number = exten for all three lines)

[striker]

hi kasthu

use a softphone to register you sip voip account in that , instead of registring int he gosutodial

and check whether your softphone is registring with your provider properly,

and also check wether u r able to make calls .

[kashutu]

Striker:

Yes i can register without Goautodial and able to receive and make calls.


William:

I have made the cahnges as you have mentioned but still getting the same error, CAUSE 20: Subscriber Abscent. What else should i look for in order to resolve the problem? I have only one carrier, however i do have five trunks from that carrier which i am planning to use once i am able to get it running.

Here is the CLI


a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Apr 18 15:21:27] Retransmitting #2 (NAT) to 10.110.1.8:30450:
INVITE sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.110.1.3:5060;branch=z9hG4bK58cfd5e5;rport
From: "S1104181521278600051" <sip:0000000000@10.110.1.3>;tag=as682e756c
To: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>
Contact: <sip:0000000000@10.110.1.3>
Call-ID: 78d4b87726009f211b0754142fe5263c@10.110.1.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S1104181521278600051" <sip:0000000000@10.110.1.3>;privacy=off;screen=no
Date: Mon, 18 Apr 2011 10:21:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 254

v=0
o=root 2362 2362 IN IP4 10.110.1.3
s=session
c=IN IP4 10.110.1.3
t=0 0
m=audio 16190 RTP/AVP 0 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Apr 18 15:21:28]
<--- SIP read from 10.110.1.8:30450 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.110.1.3:5060;branch=z9hG4bK58cfd5e5;rport=5060
To: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>
From: "S1104181521278600051" <sip:0000000000@10.110.1.3>;tag=as682e756c
Call-ID: 78d4b87726009f211b0754142fe5263c@10.110.1.3
CSeq: 102 INVITE
Content-Length: 0


<------------->
[Apr 18 15:21:28] --- (7 headers 0 lines) ---
[Apr 18 15:21:28]
<--- SIP read from 10.110.1.8:30450 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.110.1.3:5060;branch=z9hG4bK58cfd5e5;rport=5060
To: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>
From: "S1104181521278600051" <sip:0000000000@10.110.1.3>;tag=as682e756c
Call-ID: 78d4b87726009f211b0754142fe5263c@10.110.1.3
CSeq: 102 INVITE
Content-Length: 0


<------------->
[Apr 18 15:21:28] --- (7 headers 0 lines) ---
[Apr 18 15:21:28]
<--- SIP read from 10.110.1.8:30450 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.110.1.3:5060;branch=z9hG4bK58cfd5e5;rport=5060
Contact: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>
To: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>;tag=794d6c01
From: "S1104181521278600051"<sip:0000000000@10.110.1.3>;tag=as682e756c
Call-ID: 78d4b87726009f211b0754142fe5263c@10.110.1.3
CSeq: 102 INVITE
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 0


<------------->
[Apr 18 15:21:28] --- (9 headers 0 lines) ---
[Apr 18 15:21:28]
<--- SIP read from 10.110.1.8:30450 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.110.1.3:5060;branch=z9hG4bK58cfd5e5;rport=5060
Contact: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>
To: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>;tag=794d6c01
From: "S1104181521278600051"<sip:0000000000@10.110.1.3>;tag=as682e756c
Call-ID: 78d4b87726009f211b0754142fe5263c@10.110.1.3
CSeq: 102 INVITE
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 0


<------------->
[Apr 18 15:21:28] --- (9 headers 0 lines) ---
[Apr 18 15:21:29]
<--- SIP read from 10.110.1.8:30450 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.110.1.3:5060;branch=z9hG4bK58cfd5e5;rport=5060
Contact: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>
To: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>;tag=794d6c01
From: "S1104181521278600051"<sip:0000000000@10.110.1.3>;tag=as682e756c
Call-ID: 78d4b87726009f211b0754142fe5263c@10.110.1.3
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 236

v=0
o=- 8 2 IN IP4 10.110.1.8
s=CounterPath Bria Professional
c=IN IP4 10.110.1.8
t=0 0
m=audio 39628 RTP/AVP 0 96
a=fmtp:96 0-15
a=rtpmap:96 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:1D84E7B701184293AAD8FC9A1598F58A

<------------->
[Apr 18 15:21:29] --- (12 headers 10 lines) ---
[Apr 18 15:21:29] Found RTP audio format 0
[Apr 18 15:21:29] Found RTP audio format 96
[Apr 18 15:21:29] Found audio description format telephone-event for ID 96
[Apr 18 15:21:29] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Apr 18 15:21:29] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 18 15:21:29] Peer audio RTP is at port 10.110.1.8:39628
[Apr 18 15:21:29] list_route: hop: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>
[Apr 18 15:21:29] set_destination: Parsing <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on> for address/port to send to
[Apr 18 15:21:29] set_destination: set destination to 10.110.1.8, port 30450
[Apr 18 15:21:29] Transmitting (NAT) to 10.110.1.8:30450:
ACK sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.110.1.3:5060;branch=z9hG4bK756497cf;rport
From: "S1104181521278600051" <sip:0000000000@10.110.1.3>;tag=as682e756c
To: <sip:cc100@10.110.1.8:30450;rinstance=838a747662e99f4c;cpd=on>;tag=794d6c01
Contact: <sip:0000000000@10.110.1.3>
Call-ID: 78d4b87726009f211b0754142fe5263c@10.110.1.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S1104181521278600051" <sip:0000000000@10.110.1.3>;privacy=off;screen=no
Content-Length: 0


---
[Apr 18 15:21:29] > Channel SIP/cc100-00000001 was answered.
[Apr 18 15:21:29] -- Executing [8600051@default:1] MeetMe("SIP/cc100-00000001", "8600051|F") in new stack
[Apr 18 15:21:29] == Parsing '/etc/asterisk/meetme.conf': [Apr 18 15:21:29] Found
[Apr 18 15:21:29] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Apr 18 15:21:29] Found
[Apr 18 15:21:29] -- Created MeetMe conference 1023 for conference '8600051'
[Apr 18 15:21:29] -- <SIP/cc100-00000001> Playing 'conf-onlyperson' (language 'en')
[Apr 18 15:21:31] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 18 15:21:43] == Parsing '/etc/asterisk/manager.conf': [Apr 18 15:21:43] Found
[Apr 18 15:21:43] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 18 15:21:43] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-1638,2", "8600051|F") in new stack
[Apr 18 15:21:43] > Channel Local/8600051@default-1638,1 was answered.
[Apr 18 15:21:43] -- Executing [918572390065@default:1] AGI("Local/8600051@default-1638,1", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 18 15:21:43] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 18 15:21:43] -- Executing [918572390065@default:2] Dial("Local/8600051@default-1638,1", "SIP/ibrain01/18572390065||To") in new stack
[Apr 18 15:21:43] Really destroying SIP dialog '7ef11b295f2a26f1657e378911d1a578@216.24.153.216' Method: INVITE
[Apr 18 15:21:43] WARNING[5067]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Apr 18 15:21:43] == Everyone is busy/congested at this time (1:0/0/1)
[Apr 18 15:21:43] -- Executing [918572390065@default:3] Hangup("Local/8600051@default-1638,1", "") in new stack
[Apr 18 15:21:43] == Spawn extension (default, 918572390065, 3) exited non-zero on 'Local/8600051@default-1638,1'
[Apr 18 15:21:43] -- Executing [h@default:1] DeadAGI("Local/8600051@default-1638,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Apr 18 15:21:43] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 18 15:21:43] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-1638,2'
[Apr 18 15:21:43] -- Executing [h@default:1] DeadAGI("Local/8600051@default-1638,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 18 15:21:43] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 18 15:21:44]
<--- SIP read from 10.110.1.8:30450 --->



<------------->
[Apr 18 15:21:47] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 18 15:21:59] Really destroying SIP dialog '50e8aad425e1c98e337ff4577a8ace19@216.24.153.216' Method: REGISTER
ibbs*CLI> quit
Executing last minute cleanups
[root@ibbs ~]# [Apr 18 15:21:29] > Channel SIP/cc100-00000001 was answered.
-bash: [Apr: command not found
[Apr 18 15:21:29] -- Executing [8600051@default:1] MeetMe("SIP/cc100-00000001", "8600051|F") in new stack
[root@ibbs ~]# [Apr 18 15:21:29] -- Executing [8600051@default:1] MeetMe("SIP/cc100-00000001", "8600051|F") in new stack
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:29] == Parsing '/etc/asterisk/meetme.conf': [Apr 18 15:21:29] Found
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:29] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Apr 18 15:21:29] Found
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:29] -- Created MeetMe conference 1023 for conference '8600051'
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:29] -- <SIP/cc100-00000001> Playing 'conf-onlyperson' (language 'en')
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:31] == Manager 'sendcron' logged off from 127.0.0.1
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:43] == Parsing '/etc/asterisk/manager.conf': [Apr 18 15:21:43] Found
[Apr 18 15:21:43] -- Executing [h@default:1] DeadAGI("Local/8600051@default-1638,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-bash: [Apr: command not found

[root@ibbs ~]# [Apr 18 15:21:43] == Manager 'sendcron' logged on from 127.0.0.1
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:43] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-1638,2", "8600051|F") in new stack
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] > Channel Local/8600051@default-1638,1 was answered.
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:43] -- Executing [918572390065@default:1] AGI("Local/8600051@default-1638,1", "agi://127.0.0.1:4577/call_log") in new stack
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:43] -- Executing [918572390065@default:2] Dial("Local/8600051@default-1638,1", "SIP/ibrain01/18572390065||To") in new stack
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] Really destroying SIP dialog '7ef11b295f2a26f1657e378911d1a578@216.24.153.216' Method: INVITE
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:43] WARNING[5067]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] == Everyone is busy/congested at this time (1:0/0/1)
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] -- Executing [918572390065@default:3] Hangup("Local/8600051@default-1638,1", "") in new stack
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] == Spawn extension (default, 918572390065, 3) exited non-zero on 'Local/8600051@default-1638,1'
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] -- Executing [h@default:1] DeadAGI("Local/8600051@default-1638,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:43] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-1638,2'
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] -- Executing [h@default:1] DeadAGI("Local/8600051@default-1638,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-bash: syntax error near unexpected token `('
[root@ibbs ~]# [Apr 18 15:21:43] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-bash: [Apr: command not found
[root@ibbs ~]# [Apr 18 15:21:44]
-bash: [Apr: command not found
[root@ibbs ~]# <--- SIP read from 10.110.1.8:30450 --->
-bash: syntax error near unexpected token `newline'
[root@ibbs ~]#

[boybawang]

looks like you have a error in your dialplan

[kashutu]

I am dialing US with this dialplan:

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup

How can i fix it?

[williamconley]

try qualify=no

[kashutu]

qulify = no ----> didn't work either, same error


DIAL ALERT:

Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent.

[williamconley]

you may need to look up that subscriber absent.

discuss it with your provider

and check sip debug to see if there is more detail than just "subscriber absent".

[kashutu]

Issue has been resolved, thanx to Striker.

[williamconley]

Cool. What was the issue?

[kashutu]

Somehow my carrier wasn't getting registered properly. So, had to make some changes into Account Entry and Dialing plan, also had to make some changes into sip.config and extension.config.

[bryanwil]

kashutu, I am having the same problem. Would you mind be a little more specific about the changes you made to your Account Entry and Dialing plan as well as the .config files? Thank you.

[kashutu]

Hi,
What actually striker did was manually entered the account entry into sip.config file. Just try to configure your sip.config file and add the Account entry into it along with the registration string. Try it and let me know if that works for you.